Reformat the WebRTC code base
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}diff --git a/modules/audio_coding/neteq/merge.cc b/modules/audio_coding/neteq/merge.cc
index fb0bb0d..3c9ad19 100644
--- a/modules/audio_coding/neteq/merge.cc
+++ b/modules/audio_coding/neteq/merge.cc
@@ -43,10 +43,11 @@
Merge::~Merge() = default;
-size_t Merge::Process(int16_t* input, size_t input_length,
+size_t Merge::Process(int16_t* input,
+ size_t input_length,
AudioMultiVector* output) {
// TODO(hlundin): Change to an enumerator and skip assert.
- assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
+ assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
fs_hz_ == 48000);
assert(fs_hz_ <= kMaxSampleRate); // Should not be possible.
@@ -68,8 +69,8 @@
new int16_t[input_length_per_channel]);
std::unique_ptr<int16_t[]> expanded_channel(new int16_t[expanded_length]);
for (size_t channel = 0; channel < num_channels_; ++channel) {
- input_vector[channel].CopyTo(
- input_length_per_channel, 0, input_channel.get());
+ input_vector[channel].CopyTo(input_length_per_channel, 0,
+ input_channel.get());
expanded_[channel].CopyTo(expanded_length, 0, expanded_channel.get());
const int16_t new_mute_factor = std::min<int16_t>(
@@ -93,11 +94,11 @@
// Mute the new decoded data if needed (and unmute it linearly).
// This is the overlapping part of expanded_signal.
- size_t interpolation_length = std::min(
- kMaxCorrelationLength * fs_mult_,
- expanded_length - best_correlation_index);
- interpolation_length = std::min(interpolation_length,
- input_length_per_channel);
+ size_t interpolation_length =
+ std::min(kMaxCorrelationLength * fs_mult_,
+ expanded_length - best_correlation_index);
+ interpolation_length =
+ std::min(interpolation_length, input_length_per_channel);
RTC_DCHECK_LE(new_mute_factor, 16384);
int16_t mute_factor =
@@ -203,30 +204,28 @@
return required_length;
}
-int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
+int16_t Merge::SignalScaling(const int16_t* input,
+ size_t input_length,
const int16_t* expanded_signal) const {
// Adjust muting factor if new vector is more or less of the BGN energy.
const auto mod_input_length = rtc::SafeMin<size_t>(
64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
const int16_t expanded_max =
WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
- int32_t factor = (expanded_max * expanded_max) /
- (std::numeric_limits<int32_t>::max() /
- static_cast<int32_t>(mod_input_length));
+ int32_t factor =
+ (expanded_max * expanded_max) / (std::numeric_limits<int32_t>::max() /
+ static_cast<int32_t>(mod_input_length));
const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
- int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal,
- expanded_signal,
- mod_input_length,
- expanded_shift);
+ int32_t energy_expanded = WebRtcSpl_DotProductWithScale(
+ expanded_signal, expanded_signal, mod_input_length, expanded_shift);
// Calculate energy of input signal.
const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
factor = (input_max * input_max) / (std::numeric_limits<int32_t>::max() /
- static_cast<int32_t>(mod_input_length));
+ static_cast<int32_t>(mod_input_length));
const int input_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
- int32_t energy_input = WebRtcSpl_DotProductWithScale(input, input,
- mod_input_length,
- input_shift);
+ int32_t energy_input = WebRtcSpl_DotProductWithScale(
+ input, input, mod_input_length, input_shift);
// Align to the same Q-domain.
if (input_shift > expanded_shift) {
@@ -257,8 +256,10 @@
// TODO(hlundin): There are some parameter values in this method that seem
// strange. Compare with Expand::Correlation.
-void Merge::Downsample(const int16_t* input, size_t input_length,
- const int16_t* expanded_signal, size_t expanded_length) {
+void Merge::Downsample(const int16_t* input,
+ size_t input_length,
+ const int16_t* expanded_signal,
+ size_t expanded_length) {
const int16_t* filter_coefficients;
size_t num_coefficients;
int decimation_factor = fs_hz_ / 4000;
@@ -278,11 +279,10 @@
num_coefficients = 7;
}
size_t signal_offset = num_coefficients - 1;
- WebRtcSpl_DownsampleFast(&expanded_signal[signal_offset],
- expanded_length - signal_offset,
- expanded_downsampled_, kExpandDownsampLength,
- filter_coefficients, num_coefficients,
- decimation_factor, kCompensateDelay);
+ WebRtcSpl_DownsampleFast(
+ &expanded_signal[signal_offset], expanded_length - signal_offset,
+ expanded_downsampled_, kExpandDownsampLength, filter_coefficients,
+ num_coefficients, decimation_factor, kCompensateDelay);
if (input_length <= length_limit) {
// Not quite long enough, so we have to cheat a bit.
// If the input is really short, we'll just use the input length as is, and
@@ -301,15 +301,15 @@
memset(&input_downsampled_[downsamp_temp_len], 0,
sizeof(int16_t) * (kInputDownsampLength - downsamp_temp_len));
} else {
- WebRtcSpl_DownsampleFast(&input[signal_offset],
- input_length - signal_offset, input_downsampled_,
- kInputDownsampLength, filter_coefficients,
- num_coefficients, decimation_factor,
- kCompensateDelay);
+ WebRtcSpl_DownsampleFast(
+ &input[signal_offset], input_length - signal_offset, input_downsampled_,
+ kInputDownsampLength, filter_coefficients, num_coefficients,
+ decimation_factor, kCompensateDelay);
}
}
-size_t Merge::CorrelateAndPeakSearch(size_t start_position, size_t input_length,
+size_t Merge::CorrelateAndPeakSearch(size_t start_position,
+ size_t input_length,
size_t expand_period) const {
// Calculate correlation without any normalization.
const size_t max_corr_length = kMaxCorrelationLength;
@@ -328,8 +328,8 @@
new int16_t[correlation_buffer_size]);
memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
int16_t* correlation_ptr = &correlation16[pad_length];
- int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
- stop_position_downsamp);
+ int32_t max_correlation =
+ WebRtcSpl_MaxAbsValueW32(correlation, stop_position_downsamp);
int norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
correlation, norm_shift);
@@ -366,7 +366,7 @@
while (((best_correlation_index + input_length) <
(timestamps_per_call_ + expand_->overlap_length())) ||
((best_correlation_index + input_length) < start_position)) {
- assert(false); // Should never happen.
+ assert(false); // Should never happen.
best_correlation_index += expand_period; // Jump one lag ahead.
}
return best_correlation_index;
@@ -376,5 +376,4 @@
return fs_hz_ / 100 * num_channels_; // 10 ms.
}
-
} // namespace webrtc